Gradio WebRTC ⚡️
Stream video and audio in real time with Gradio using WebRTC.
## Installation
```bash
gradio cc install
gradio cc build --no-generate-docs
```
```bash
pip install dist/gradio_webrtc-0.0.30.dev0-py3-none-any.whl
```
## Docs
https://freddyaboulton.github.io/gradio-webrtc/
## Examples
```python
import asyncio
import base64
from io import BytesIO
import gradio as gr
import numpy as np
from gradio_webrtc import (
AsyncAudioVideoStreamHandler,
WebRTC,
VideoEmitType,
AudioEmitType,
)
from PIL import Image
def encode_audio(data: np.ndarray) -> dict:
"""Encode Audio data to send to the server"""
return {"mime_type": "audio/pcm", "data": base64.b64encode(data.tobytes()).decode("UTF-8")}
def encode_image(data: np.ndarray) -> dict:
with BytesIO() as output_bytes:
pil_image = Image.fromarray(data)
pil_image.save(output_bytes, "JPEG")
bytes_data = output_bytes.getvalue()
base64_str = str(base64.b64encode(bytes_data), "utf-8")
return {"mime_type": "image/jpeg", "data": base64_str}
class VideoChatHandler(AsyncAudioVideoStreamHandler):
def __init__(
self, expected_layout="mono", output_sample_rate=24000, output_frame_size=480
) -> None:
super().__init__(
expected_layout,
output_sample_rate,
output_frame_size,
input_sample_rate=24000,
)
self.audio_queue = asyncio.Queue()
self.video_queue = asyncio.Queue()
self.quit = asyncio.Event()
self.session = None
self.last_frame_time = 0
def copy(self) -> "VideoChatHandler":
return VideoChatHandler(
expected_layout=self.expected_layout,
output_sample_rate=self.output_sample_rate,
output_frame_size=self.output_frame_size,
)
async def video_receive(self, frame: np.ndarray):
newFrame = np.array(frame)
newFrame[0:, :, 0] = 255 - newFrame[0:, :, 0]
self.video_queue.put_nowait(newFrame)
async def video_emit(self) -> VideoEmitType:
return await self.video_queue.get()
async def receive(self, frame: tuple[int, np.ndarray]) -> None:
frame_size, array = frame
self.audio_queue.put_nowait(array)
async def emit(self) -> AudioEmitType:
if not self.args_set.is_set():
await self.wait_for_args()
array = await self.audio_queue.get()
return (self.output_sample_rate, array)
def shutdown(self) -> None:
self.quit.set()
self.connection = None
self.args_set.clear()
self.quit.clear()
css = """
footer {
display: none !important;
}
"""
with gr.Blocks(css=css) as demo:
webrtc = WebRTC(
label="Video Chat",
modality="audio-video",
mode="send-receive",
video_chat=True,
elem_id="video-source",
)
webrtc.stream(
VideoChatHandler(),
inputs=[webrtc],
outputs=[webrtc],
time_limit=150,
concurrency_limit=2,
)
if __name__ == "__main__":
demo.launch()
```
## Deployment
When deploying in a cloud environment (like Hugging Face Spaces, EC2, etc), you need to set up a TURN server to relay the WebRTC traffic.
The easiest way to do this is to use a service like Twilio.
```python
from twilio.rest import Client
import os
account_sid = os.environ.get("TWILIO_ACCOUNT_SID")
auth_token = os.environ.get("TWILIO_AUTH_TOKEN")
client = Client(account_sid, auth_token)
token = client.tokens.create()
rtc_configuration = {
"iceServers": token.ice_servers,
"iceTransportPolicy": "relay",
}
with gr.Blocks() as demo:
...
rtc = WebRTC(rtc_configuration=rtc_configuration, ...)
...
```
## Contributors
[csxh47](https://github.com/xhup)
[bingochaos](https://github.com/bingochaos)
[sudowind](https://github.com/sudowind)
[emililykimura](https://github.com/emililykimura)
[Tony](https://github.com/raidios)
[Cheng Gang](https://github.com/lovepope)
